AudioCodes Mediant 500 base Enterprise Session Border Controller and Media Gateway M500-ESBC
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AudioCodes Mediant 500 base Enterprise Session Border Controller and Media Gateway M500-ESBC

Available:In Stock
£518.04
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AudioCodes Mediant 500 base Enterprise Session Border Controller and Media Gateway M500-ESBC

Please Note: Additions including sessions, remote implementation support and licenses/ software are available. If you require any of these additions, please enquire using the button above.

The AudioCodes Mediant 500 is a compact, high-performance VoIP connectivity solution for small enterprises and branch office locations.

AudioCodes Mediant 500 Features 

  • 250 SBC Sessions
  • 30 TDM Sessions
  • 1+1 High Availability
  • WebRTC Gateway
  • Comprehensive interoperability, proven interoperability with SIP trunks, SIP platforms and IP cloud services
  • Hybrid functionality, true hybrid SBC and gateway platform for gradual migration, low CAPEX and reduced space and power footprints
  • Enhanced security, robust perimeter defence against cyber, DoS and DDoS attacks, as well as eavesdropping, fraud and service theft
  • Superior voice quality, advanced capabilities for optimizing and monitoring voice service quality
  • High resiliency, local branch survivability and PSTN fallback with E911

Scaling up to 250 concurrent sessions, the Mediant 500 connects IP-PBXs to any SIP trunking service provider and offers superior performance in connecting any SIP to SIP environment.


The Mediant 500 also supports up to 30 voice channels in a 1U platform to enable versatile connectivity between TDM and VoIP networks, such as connecting legacy TDM PBX systems to IP networks and IP-PBXs to the PSTN.

AudioCodes Mediant 500 Technical Specification

Capacities

  • Max. Signaling: 250
  • Max. Registered Users: 1,500
  • Max. RTP/SRTP Sessions: 200

Telephony Interfaces

  • Digital: Single E1/T1 interface
  • Analog: 5 ppm High Precision
  • Clock Source: Various ISDN PRI protocols such as EuroISDN, North American NI-2, LucentTM 4/5ESS, NortelTM DMS- 100 and others. Different CAS protocols, including MFC R2,

Network Interfaces

  • Ethernet: 4 GE interfaces configured in 1+1 redundancy or as individual ports

Security

  • Access Control: DoS/DDoS line rate protection, bandwidth throttling, dynamic blacklisting (Intrusion Detection System)
  • VoIP Firewall: RTP pinhole management, rogue RTP detection and prevention, SIP message policy, advanced RTP latching
  • Encryption/Authentication: TLS, SRTP, HTTPS, SSH, client/server SIP Digest authentication, RADIUS Digest
  • Privacy: Topology hiding, user privacy
  • Traffic Separation: VLAN/physical interface separation for multiple media, control and OAMP interfaces

Interoperability

  • SIP B2BUA: Full SIP transparency, mature and broadly deployed SIP stack, stateful proxy mode
  • SIP Interworking: 3xx redirect, REFER, PRACK, session timer, early media, call hold, delayed offer
  • Registration and Authentication: User registration restriction control, registration and authentication on behalf of users, SIP authentication server for SBC users
  • Transport Mediation: Mediation between SIP over UDP/TCP/TLS, IPv4/IPv6, RTP/SRTP (SDES)
  • Header Manipulation: Add/modify/delete SIP headers and message body using simple WireShark-like language with powerful capabilities such as variables and utility functions
  • Number Manipulations: Ingress and egress digit manipulation
  • SIP Interworking: 3xx redirect, REFER, PRACK, session timer, early media, call hold, delayed offer
  • Signal Conversion: DTMF/RFC 2833/SIP, T.38 fax, T.38 V3, packet-time conversion
  • NAT: Local and far-end NAT traversal for support of remote workers

Voice Quality and SLA

  • Call Admission Control: Limit number and rate of concurrent sessions and registers per peer for inbound and outbound directions
  • Packet Marking: 802.1p/Q VLAN tagging, DiffServ, TOS
  • Standalone Survivability: Maintains local calls in the event of WAN failure. Outbound calls can use PSTN fallback (including E911).
  • Voice Monitoring and Enhancement: Transrating, RTCP-XR, acoustic echo cancellation, replacing voice profile due to impairment detection, fixed and dynamic voice gain control, packet loss concealment, dynamic programmable jitter buffer, silence suppression/comfort noise generation, RTP redundancy, broken connection detection
  • Direct Media: Hair-pinning (no media anchoring) of local calls to avoid unnecessary media delays and bandwidth consumption
  • Test Agent: Ability to remotely verify connectivity, voice quality and SIP message flow between SIP UAs
  • High Availability SBC high availability with two-box redundancy, active calls preserved

SIP Routing

  • Routing Criteria Incoming SIP trunk, DID ranges, host names, any SIP headers, codecs, QoE, bandwidth
  • Querying External Databases Routing based on customized queries of ENUM, LDAP, HTTP server (REST API)
  • Route To Configured SIP peers, registered users, IP address, request URI
  • Advanced Routing Features Alternative routes, load balancing, least-cost routing, call forking, E911 emergency call detection and prioritisation
  • SIPREC IETF standard SIP recording interface

Management

  • OAM&P Browser-based GUI, CLI, SNMP, INI Configuration file, REST API, EMS

Physical and Environmental

  • Dimensions 43.7 (1U) x 310 x 210 mm (HxWxD)
  • Weight 4.4 lb (2.0kg)
  • Mounting Desktop or 19” rack mount
  • Power 100-240V, 50-60 Hz, 0.8A
  • Environmental
  • Operational: 0 to 40°C (41 to 104°F); Storage: -25 to 70°C (-13 to 185°F) Relative Humidity: 10 to 90% non-condensing

AudioCodes Mediant 500 base Enterprise Session Border Controller and Media Gateway M500-ESBC

Please Note: Additions including sessions, remote implementation support and licenses/ software are available. If you require any of these additions, please enquire using the button above.

The AudioCodes Mediant 500 is a compact, high-performance VoIP connectivity solution for small enterprises and branch office locations.

AudioCodes Mediant 500 Features 

  • 250 SBC Sessions
  • 30 TDM Sessions
  • 1+1 High Availability
  • WebRTC Gateway
  • Comprehensive interoperability, proven interoperability with SIP trunks, SIP platforms and IP cloud services
  • Hybrid functionality, true hybrid SBC and gateway platform for gradual migration, low CAPEX and reduced space and power footprints
  • Enhanced security, robust perimeter defence against cyber, DoS and DDoS attacks, as well as eavesdropping, fraud and service theft
  • Superior voice quality, advanced capabilities for optimizing and monitoring voice service quality
  • High resiliency, local branch survivability and PSTN fallback with E911

Scaling up to 250 concurrent sessions, the Mediant 500 connects IP-PBXs to any SIP trunking service provider and offers superior performance in connecting any SIP to SIP environment.


The Mediant 500 also supports up to 30 voice channels in a 1U platform to enable versatile connectivity between TDM and VoIP networks, such as connecting legacy TDM PBX systems to IP networks and IP-PBXs to the PSTN.

AudioCodes Mediant 500 Technical Specification

Capacities

  • Max. Signaling: 250
  • Max. Registered Users: 1,500
  • Max. RTP/SRTP Sessions: 200

Telephony Interfaces

  • Digital: Single E1/T1 interface
  • Analog: 5 ppm High Precision
  • Clock Source: Various ISDN PRI protocols such as EuroISDN, North American NI-2, LucentTM 4/5ESS, NortelTM DMS- 100 and others. Different CAS protocols, including MFC R2,

Network Interfaces

  • Ethernet: 4 GE interfaces configured in 1+1 redundancy or as individual ports

Security

  • Access Control: DoS/DDoS line rate protection, bandwidth throttling, dynamic blacklisting (Intrusion Detection System)
  • VoIP Firewall: RTP pinhole management, rogue RTP detection and prevention, SIP message policy, advanced RTP latching
  • Encryption/Authentication: TLS, SRTP, HTTPS, SSH, client/server SIP Digest authentication, RADIUS Digest
  • Privacy: Topology hiding, user privacy
  • Traffic Separation: VLAN/physical interface separation for multiple media, control and OAMP interfaces

Interoperability

  • SIP B2BUA: Full SIP transparency, mature and broadly deployed SIP stack, stateful proxy mode
  • SIP Interworking: 3xx redirect, REFER, PRACK, session timer, early media, call hold, delayed offer
  • Registration and Authentication: User registration restriction control, registration and authentication on behalf of users, SIP authentication server for SBC users
  • Transport Mediation: Mediation between SIP over UDP/TCP/TLS, IPv4/IPv6, RTP/SRTP (SDES)
  • Header Manipulation: Add/modify/delete SIP headers and message body using simple WireShark-like language with powerful capabilities such as variables and utility functions
  • Number Manipulations: Ingress and egress digit manipulation
  • SIP Interworking: 3xx redirect, REFER, PRACK, session timer, early media, call hold, delayed offer
  • Signal Conversion: DTMF/RFC 2833/SIP, T.38 fax, T.38 V3, packet-time conversion
  • NAT: Local and far-end NAT traversal for support of remote workers

Voice Quality and SLA

  • Call Admission Control: Limit number and rate of concurrent sessions and registers per peer for inbound and outbound directions
  • Packet Marking: 802.1p/Q VLAN tagging, DiffServ, TOS
  • Standalone Survivability: Maintains local calls in the event of WAN failure. Outbound calls can use PSTN fallback (including E911).
  • Voice Monitoring and Enhancement: Transrating, RTCP-XR, acoustic echo cancellation, replacing voice profile due to impairment detection, fixed and dynamic voice gain control, packet loss concealment, dynamic programmable jitter buffer, silence suppression/comfort noise generation, RTP redundancy, broken connection detection
  • Direct Media: Hair-pinning (no media anchoring) of local calls to avoid unnecessary media delays and bandwidth consumption
  • Test Agent: Ability to remotely verify connectivity, voice quality and SIP message flow between SIP UAs
  • High Availability SBC high availability with two-box redundancy, active calls preserved

SIP Routing

  • Routing Criteria Incoming SIP trunk, DID ranges, host names, any SIP headers, codecs, QoE, bandwidth
  • Querying External Databases Routing based on customized queries of ENUM, LDAP, HTTP server (REST API)
  • Route To Configured SIP peers, registered users, IP address, request URI
  • Advanced Routing Features Alternative routes, load balancing, least-cost routing, call forking, E911 emergency call detection and prioritisation
  • SIPREC IETF standard SIP recording interface

Management

  • OAM&P Browser-based GUI, CLI, SNMP, INI Configuration file, REST API, EMS

Physical and Environmental

  • Dimensions 43.7 (1U) x 310 x 210 mm (HxWxD)
  • Weight 4.4 lb (2.0kg)
  • Mounting Desktop or 19” rack mount
  • Power 100-240V, 50-60 Hz, 0.8A
  • Environmental
  • Operational: 0 to 40°C (41 to 104°F); Storage: -25 to 70°C (-13 to 185°F) Relative Humidity: 10 to 90% non-condensing

Understanding Our Returns and Refunds Policy

Easy Returns for Your Convenience

At our company, we believe that shopping should be hassle-free. This is why we have made our returns and refunds process as easy as possible. If you receive a product that doesn’t meet your expectations or is defective, you can return it within 30 days for a full refund. Our team is always here to assist you, so you don’t have to worry about complicated steps when returning an item.

Refund Process Made Simple

Once we receive your returned item, we initiate the refund process immediately. Typically, refunds are processed within 5-7 business days. However, there might be variations due to your bank's policies. We understand that getting your money back promptly is essential, which is why we prioritize quick refunds. Because of this efficiency, customers can trust that their satisfaction is our highest priority.

Customer Support for Returns and Refunds

If you have questions about our returns and refunds, our customer support team is always ready to help. You can reach out via email or phone for assistance with your return. We want you to feel confident when making a purchase, and knowing that we value customer support makes the process smoother. So, don’t hesitate to contact us if you need clarification or help!